Hello all,
Reading the file produced by 'alsactl store', I learn
that my sound hardware has a number of control parameters
that have names, types, values, ranges, etc. etc.
I now want to write some hopefully not too convolved
C or C++ code to read and write these parameters.
Is there, after X years of ALSA, any documentation that
explains the basic concepts and tells me how to do this ?
If such a thing exists I can't find it.
The Doxygen info on the ALSA site is completely useless
for the purpose of learning to understand and use the
control interface.
The textual information provided there usually provides
*nothing* that can't be read from the C types, structs or
functions it is supposed to document. It just repeats the
jargon used in the code, and is at least 99.9% redundant.
What these things actually mean, how they fit together
and what is the big picture is AFAIK nowhere and never
explained. Which is strange, because if you design a
system such as this, that would be the absolutely first
thing you need to define. No doubt the designers have it
in their heads. No doubt it's well structured and also
abstracted to almost absurd levels. But it remains a
complete mystery unless you have the time and energy and
someone is paying you to spend at least half a year to
reverse-engineer the so-called docs. If ever there was
an example of Doxygen or similar system being no more
than a pretext to keep the quality department happy,
ALSA is the best one I know of.
Now if someone can point me to some existing docs that
explain how I can e.g. set the sample clock source on a
RME MADI card in less than ten lines of C code (knowing
the parameter names, ranges, etc - no need to find them
out dynamically, I can read them asound.state) then I'll
eat my hat. It shouldn't be difficult. On some competing
systems all it takes is one ioctl().
Ciao,
--
FA
Laboratorio di Acustica ed Elettroacustica
Parma, Italia
Lascia la spina, cogli la rosa.
Hi,
I can't find anything online that gives me a way to run /sbin/mkdosfs as
a normal user.
Is it just that I need to add the user to the mkdosfs group or something
similar?
Cheers.
--
Patrick Shirkey
Boost Hardware Ltd
Hello folks!
One question, I hope it's not too dumb. :-(
If you have your average patchbay, how does it know, when new MIDI/audio
ports/clients come to live or die? And how does it know, that some connection
was killed by some other application.
Does it simply query it all the time? I wouldn't think so... But perhaps I'm
wrong...
Thanks for hints on this.
Kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
Ok, things have settled down, and i've tweaked a little here and there.
Seems to be running nicely now, and fairly stable.
A screenshot. of a generic setup.
http://shup.com/Shup/81262/patchage3.png
Alex.
lpatchage
jackdbus
rosegarden
linuxsampler
ardour2
jconv
On Tue, Nov 11, 2008 at 10:35 PM, alex stone <compose59(a)gmail.com> wrote:
> Nedko,
>
> This is what i get when i try, in the messages window of lpatchage, when i
> try to connect linuxsampler audio out:
>
> [JACKDBUS] ConnectPortsByName() failed.
>
> jackdbus log is attached. (I've renamed a copy for your perusal)
>
> Alex.
>
>
>
>
> On Tue, Nov 11, 2008 at 8:55 PM, Nedko Arnaudov <nedko(a)arnaudov.name>wrote:
>
>> "alex stone" <compose59(a)gmail.com> writes:
>>
>> > But i'm still at a loss as to why i can't connect LS audio out, to
>> Ardour
>> > audio in, in lpatchage, visibly.
>> > It works in Qjackctl, but stubbornly refuses to connect in lpatchage,
>> even
>> > though the actual connections are made in Ardour, and most importantly,
>> > work.
>>
>> Do you get any errors in jackdbus log file when you are trying to
>> connect using lpatchage?
>>
>> --
>> Nedko Arnaudov <GnuPG KeyID: DE1716B0>
>>
>
>
release candidate 2 has some important fixes:
* Fix for #46 - on first save of newly appeared clients, their state
was not correcttly recorded as being saved and thus was not being
restored on project load afterwards.
* Memory corruption fixes caused by bug in stdout/stderr handling
code. Was happening when lash client outputs lot of data to stdout or
stderr
* Improved handling of repeating lines sent to stdout/stderr
I would like to ask LASH beleivers and other interested parties to test
the 0.6.0 release candidate. Juuso Alasuutari and me have been doing
some major changes to the lash code. We have done lot of work, we've
fixed several big implementation issues and we need stable point before
doing more changes (0.6.1 and 1.0 milestones).
In the tarball there is simple lash_control script. One can also control
LASH through patchage-0.4.2 and through lpatchage (availabe through
git).
User visible changes since 0.5.4:
* Use jack D-Bus interface instead of libjack, enabled by default, can
be disabled. Ticket #1
* Allow controlling LASH through D-Bus. Ticket #2
* Use D-Bus autolaunching instead of old mechanism. Ticket #3
* Log file (~/.log/lash/lash.log) for LASH daemon. Ticket #4
* Client stdout/stderr are logged to lash.log, when clients are
launched by LASH daemon (project restore). Ticket #5
* Improved handling of misbehaved clients. Ticket #45
* Projects now can have comment and notes associated. Ticket #13
Download:
http://download.savannah.gnu.org/releases/lash/lash-0.6.0~rc2.tar.bz2http://download.savannah.gnu.org/releases/lash/lash-0.6.0~rc2.tar.bz2.sig
--
Nedko Arnaudov <GnuPG KeyID: DE1716B0>
For a new audio application I need to code a JACK client with C++. So
far I did it only with C and have a problem with giving the pointer to
the callback process function, which is a method now. So what is the
best performing solution? Is a delegate function a good idea, being
static and triggering the method in the objectinstance?
Cheers,
Malte
--
----
media art + development
http://www.block4.com
current events:
exhibition spame-moi La Motte-Servolex, France 17.10.-20.12.2008
Hi all,
I need to use a microphone input as a trigger. In other words my idea is
to connect a switch to the microphone input. In this way, when the
switch is turned on it generates a spike in the captured track.
I would like to create a program that trigger an event every spike it
receives.
I succeed in capturing the mic input through a simple program that uses
alsa driver, but I don't know how to "parse" the raw data to search for
the spikes. Any hints?
Second question: on a "full duplex" sound card, can I capture at 8 bit,
mono, 22.050 bit/s , and on the same time playback at 16 bit, stereo,
44.100 ?
Thank you!
Lorenzo
I'm a home studio enthusiast and also a former JAVA programmer.
I'm looking at combining these interests and contributing to a
project, but most apps seem to be written in C++.
Any suggestions? Anyone involved in any good JAVA projects?
--
Cheers, Craig
http://craiglawton.infohttp://romansandals.wordpress.com
Hi,
I have a soundcard M Audio Fast Track Pro 4x4 and I'm trying use four outputs at the same time, but I have
no sucess.
I tried configure with the tips of alsa-project page and here its instructions don't work correctly.
Any suggestions?
Guilherme Bertissolo
Novos endereços, o Yahoo! que você conhece. Crie um email novo com a sua cara @ymail.com ou @rocketmail.com.
http://br.new.mail.yahoo.com/addresses
